My iax.conf is: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=100 jittershrinkrate=1 notransfer=yes trunk=no
[zoot] type=user secret=xxxxxxxx auth=plaintext host=zoot.xxxxxxxxxx.net notransfer=yes context=from-trunk On 11/8/06, Matt <[EMAIL PROTECTED]> wrote:
I (and some others) are having an issue with placing calls on hold. Our setup is as follows: IAX2 or SIP terminator/originator ---> asterisk box ---> SIP Phones I have tried asterisk verison 1.0.9, 1.2.6 and 1.2.12, and all have the same issue. When I place a call on hold that has come in a PSTN channel (through a PRI and a Digium card) everything is fine. When I place a call on hold that has come in or gone out an IAX2 or SIP terminator or originator, when I pick the call back up often there is one-way-audio (I can hear the caller, but the caller can not hear me). I have attached a packet capture of the situation, and as you can see at about packet #3803 the audio goes one way. Can anyone enlighten me as to why this is happening, and why asterisk is no longer sending audio back to the terminator? I would like to get this fixed, obviously. (This file was a tcpdump, and can be opened in ethereal/wireshark) EDIT: Packet capture at following address: http://hecate.chilitech.net/~matth/zootdump.cap
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